Abstract. Avaya Solution & Interoperability Test Lab

Transcription

1 Avaya Solution & Interoperability Test Lab Application Notes for Configuring TELUS SIP Trunking with Avaya Aura Communication Manager and Acme Packet 3800 Net-Net Session Border Controller Issue 1.0 Abstract These Application Notes describe the steps to configure Session Initiation Protocol (SIP) Trunking between TELUS SIP Trunking and an Avaya SIP- enterprise solution. The Avaya solution consists of Avaya Aura Communication Manager 5.2.1, Acme Packet 3800 Net-Net Session Border Controller and various Avaya endpoints. TELUS is a member of the Avaya DevConnect Service Provider program. Information in these Application Notes has been obtained through DevConnect compliance testing and additional technical discussions. Testing was conducted via the DevConnect Program at the Avaya Solution and Interoperability Test Lab. 1 of 60

2 1. Introduction These Application Notes describe the steps to configure Session Initiation Protocol (SIP) Trunking between TELUS SIP Trunking and an Avaya SIP- enterprise solution. The Avaya solution consists of Avaya Aura Communication Manager 5.2.1, Acme Packet 3800 Net-Net Session Border Controller and various Avaya endpoints. Customers using this Avaya SIP- enterprise solution with TELUS SIP Trunking are able to place and receive PSTN calls via a broadband WAN connection and the SIP protocol. This converged network solution is an alternative to traditional PSTN trunks such as ISDN-PRI. 2. General Test Approach and Test Results The general test approach was to connect a simulated enterprise site to TELUS SIP Trunking via the public Internet and exercise the features and functionality listed in Section 2.1. The simulated enterprise site was comprised of Communication Manager and the Acme Packet 3800 Net-Net SBC. DevConnect Compliance Testing is conducted jointly by Avaya and DevConnect members. The jointly-defined test plan focuses on exercising APIs and/or standards-based interfaces pertinent to the interoperability of the tested products and their functionalities. DevConnect Compliance Testing is not intended to substitute full product performance or feature testing performed by DevConnect members, nor is it to be construed as an endorsement by Avaya of the suitability or completeness of a DevConnect member s solution Interoperability Compliance Testing To verify SIP trunking interoperability, the following features and functionality were covered during the interoperability compliance test. Please note that enterprise SIP endpoints are not supported in this configuration and so were not tested. Response to SIP OPTIONS queries Incoming PSTN calls to various enterprise phone types including H.323, digital, and analog telephones. All inbound PSTN calls were routed to the enterprise across the SIP trunk from the service provider. Outgoing PSTN calls from various enterprise phone types including H.323, digital, and analog telephones at the enterprise. All outbound PSTN calls were routed from the enterprise across the SIP trunk to the service provider. Inbound and outbound PSTN calls to/from Avaya one-x Communicator (soft client) Avaya one-x Communicator can place calls from the local computer or control a remote phone. Both of these modes were tested. Avaya one-x Communicator also supports two Voice Over IP (VoIP) protocols: H.323 and SIP. Only the H.323 version of Avaya one-x Communicator was tested. Inbound and outbound calls to/from TELUS Derived Voice endpoints Inbound and outbound calls to/from TELUS Mobility endpoints 2 of 60

3 Various call types including: local, long distance, international, outbound toll-free, operator assisted calls, and local directory assistance (411). Codec G.711MU, G.711A and G.729A. DTMF transmission using RFC 2833 Caller ID presentation and Caller ID restriction Response to incomplete call attempts and trunk errors. Voic navigation for inbound and outbound calls Voic Message Waiting Indicator (MWI) User features such as hold and resume, internal call forwarding, transfer, and conference Off-net call forwarding and enterprise mobility (extension to cellular) T.38 Fax (established from an initial G.711MU/A call) Network Call Redirection using the SIP REFER method Items not supported or not tested included the following: Inbound toll-free and emergency calls are supported but were not tested. Call redirection requested by a 302 response is not supported by TELUS. Establishment of a T.38 fax from a G.729 call could not be tested due to a limitation of the lab environment Test Results Interoperability testing of TELUS SIP Trunking was completed with successful results for all test cases with the exception of the observations/limitations described below. OPTIONS Max-Forwards Value: TELUS requires that SIP OPTIONS messages sent from the enterprise contain a Max-Forwards value of zero. The SBC was configured to originate OPTIONS messages to the network with Max-Forward set to zero. See the ping-method setting in Section Use of SA8965: TELUS requires re-invites to contain Session Description Protocol (SDP) information. Thus, the Communication Manager special application SA8965 must be. (See Section 5.2) Incorrect Error Indication if No Matching Codec Offered: If the Communication Manager SIP trunk is improperly configured to have no matching codec with the service provider and an inbound call is placed, Communication Manager returns a 500 Service Unavailable response instead of a 488 Not Acceptable Here response. The user hears a network announcement. Re-INVITE When Covering to Voic Does Not Contain SDP: In the scenario of an inbound call to an enterprise extension that covers to voic , the resulting re- INVITE to redirect media between the SBC/network and the voic server does not contain SDP. This seems to be an issue with special application SA8965 and is being investigated by Avaya development. This scenario did not cause any user perceivable problems. Voic messages could be left by the caller and retrieved by the recipient. Codecs: G.711MU is the preferred/default codec used by TELUS. G.729 is supported but is not available on all media gateways in the TELUS network. 3 of 60

4 Call Fowarding and EC500: Inbound PSTN calls that are call forwarded back to the PSTN or ring to an EC500 (enterprise mobility) endpoint, will display the forwarding party/ec500 host at the destination instead of the original PSTN caller. This is the result of differences in the interpretation/implementation of the SIP Diversion header between TELUS and Communication Manager. A SIP header manipulation was created on the SBC to create the P-Asserted-ID header with information contained in the Diversion header. (See Sections and ) This allows the call to complete but results in the incorrect calling party displayed at the destination as described above. Calling Party Number (PSTN transfers): The calling party number displayed on the PSTN phone is not updated to reflect the true connected party on calls that are transferred to the PSTN. After the call transfer is complete, the calling party number displays the number of the transferring party and not the actual connected party. Communication Manager provides the new connected party information by updating the Contact header in a re-invite message. TELUS does not use the updated Contact header for displaying calling party information. Coverage to Voic for TELUS Mobility Users: Calls from the enterprise to TELUS mobility users that cover to voic could result in one-way audio. If this occurs, the caller will not be able to hear the voic announcements and menus. A software change was made on Communication Manager to address this issue and was built on top of Release Service Pack 11. The change was tested and passed compliance testing using an early development release. The change will be available in a future service pack release. Customers who encounter this problem should use the standard escalation process to request a patch from Avaya Global Services. Use of REFER: Enabling of the Network Call Redirection feature on the Communication Manager SIP trunk activates the use of the SIP REFER method for various inbound PSTN calls redirected back to the PSTN. The use of the REFER method resulted in dropped calls for blind transfer and vector redirection scenarios. Enabling of Network Call Redirection is not recommended. T.38 Fax Network Coverage: Not all media gateways in the TELUS network support T.38 fax. Communication Manager does not support fallback to G.711 pass-through fax from T.38 fax. Thus, if a T.38 fax call encounters a media gateway in the TELUS network that does not support T.38 then the call will terminate. Transitioning to T.38 for Outbound Calls: In general, the answering side of a fax call will send a re-invite to transition to T.38. For outbound fax calls to the PSTN, this means the network would typically send the re-invite to transition to T.38. However, TELUS requires Communication Manager to transition to T.38 for both inbound and outbound fax calls. Relying on Communication Manager to transition to T.38 on an outbound call may have the following impact: On an outbound call, sending of the T.38 INVITE happens on detection of the V.21 preamble of the originating fax machine s Digital Command Signal (DCS) message. This is part of the T.30 exchange. This request to transition to T.38 may happen too late for some terminating gateways to accommodate the switch to T.38. If the initial call is using the G.729 codec, the compression of the V.21 preamble may cause its detection to be less reliable then if the call was initially using G of 60

5 The ability to transition to T.38 in the middle of the T.30 exchange is supported on the following Avaya media platforms (G430/G450/TN2602). Older platforms (G350/G700/TN2302) may have different behavior. Compliance testing was conducted with the TN2602 media platform (part of the G650 media gateway) using codec G.711MU to initially establish the call. Outbound T.38 fax calls in this environment were successful. G.711 Pass-through Fax: Communication Manager does not support G.711 passthrough fax over SIP trunks. These calls are treated like any other voice call by Communication Manager. If a customer chooses to use G.711 pass-through fax, success is not guaranteed. Inbound Calls With Calling Party Number (CPN) Block Enabled: Communication Manager will accept calls from SIP domains/hosts that it recognizes. This is determined by comparing the uri-host portion of the P-Asserted-Identity (PAI) header or From header in the incoming INVITE with the Far-end domain configured in the Communication Manager signaling group. If Communication Manager finds a match in any signaling group, the call is accepted. TELUS does not include the PAI header in the INVITE and in the case of CPN block, the uri-host in the From header is masked with the value of anonymous.invalid. In order to allow these calls to complete, a second signaling/trunk group was created on Communication Manager to accept calls from domain anonymous.invalid. See Sections 5.7 and 5.8 for details Support For technical support on the TELUS system, please contact your TELUS Account Executive or visit Avaya customers may obtain documentation and support for Avaya products by visiting Selecting the Support Contact Options link followed by Maintenance Support provides the worldwide support directory for Avaya Global Services. Specific numbers are provided for both customers and partners based on the specific type of support or consultation services needed. Some services may require specific Avaya service support agreements. Alternatively, in the United States, (866) GO-AVAYA ( ) provides access to overall sales and service support menus. 3. Reference Configuration Figure 1 illustrates a sample Avaya SIP- enterprise solution connected to TELUS SIP Trunking. This is the configuration used for compliance testing. The Avaya components used to create the simulated customer site included: Duplex Avaya S8710 Servers running Communication Manager Avaya G650 Media Gateway Avaya 9600-Series IP telephones (H.323) Avaya 4600-Series IP telephones (H.323) Avaya 1600-Series IP telephones (H.323) 5 of 60

6 Avaya one-x Communicator (H.323) Avaya digital and analog telephones Located at the edge of the enterprise is the 3800 Net-Net SBC. It has a public side that connects to the external network and a private side that connects to the enterprise network. All SIP and RTP traffic entering or leaving the enterprise flows through the 3800 Net-Net SBC. In this way, the 3800 Net-Net SBC can protect the enterprise against any SIP-based attacks. The 3800 Net- Net SBC provides network address translation at both the IP and SIP layers. For security reasons, any actual public IP addresses used in the configuration have been replaced with private IP addresses. Similarly, any references to real routable PSTN numbers have also been changed to numbers that can not be routed by the PSTN. Figure 1: Avaya IP Telephony Network using TELUS SIP Trunking 6 of 60

7 For inbound calls, the calls flow from the service provider to the 3800 Net-Net SBC then to Communication Manager. Once the call arrives at Communication Manager, incoming call treatment, such as incoming digit translations and class of service restrictions may be performed. Outbound calls to the PSTN are first processed by Communication Manager and may be subject to outbound features such as automatic route selection, digit manipulation and class of service restrictions. Once Communication Manager selects the proper SIP trunk, the call is routed to the 3800 Net-Net SBC. From the 3800 Net-Net SBC, the call is sent to TELUS SIP Trunking. For outbound calls, the enterprise was configured to send 11 digits in the SIP destination headers (Request URI and To) and 10 digits in the SIP source headers (i.e., From, Contact, and P- Asserted-Identity). For inbound calls, TELUS sent 10 digits in both the source headers and destination headers. 4. Equipment and Software Validated The following equipment and software were used for the sample configuration provided: Avaya IP Telephony Solution Components Component Release Avaya Aura Communication Manager running on duplex Avaya S8710 Servers SP11 + dev patch (R015x ) + dev patch Avaya G650 Media Gateway IP Server Interface (IPSI) TN2312BP HW15 FW054 Control LAN (CLAN) TN799DP HW01 FW040 IP Media Processor (MEDPRO) TN2602AP HW02 FW061 Avaya 1608 IP Telephone (H.323) Avaya one-x Deskphone Value Edition Avaya 4621SW IP Telephone (H.323) Avaya 9640 IP Telephone (H.323) Avaya one-x Deskphone Edition 3.1 SP4 (3.1.04S) Avaya one-x Communicator (H.323) 6.1 (Build GA-30334) Avaya 2420 Digital Telephone n/a Avaya 6210 Analog Telephone n/a Acme Packet 3800 Net-Net Session Border SCX6.2.0 MR-3 GA (Build 619) Controller TELUS SIP Trunking Solution Components Component Release Acme Packet 4520 Net-Net Session Border Controller 6.1m7p5 Nokia Siemens Networks HiQ 4200 Version 14.0 Table 1: Equipment and Software Tested 7 of 60

8 The specific configuration above was used for the compliance testing. Note that this solution will be compatible with other Avaya Server and Media Gateway platforms running similar versions of Communication Manager. 5. Configure Avaya Aura Communication Manager This section describes the procedure for configuring Communication Manager for TELUS SIP Trunking. A SIP trunk is established between Communication Manager and the 3800 Net-Net SBC for use by traffic to and from TELUS. It is assumed the general installation of Communication Manager, Avaya Media Gateway and 3800 Net-Net SBC has been previously completed and is not discussed here. The Communication Manager configuration was performed using the System Access Terminal (SAT). Some screens in this section have been abridged and highlighted for brevity and clarity in presentation. Note that the IP addresses and phone numbers shown throughout these Application Notes have been edited so that the actual public IP addresses of the network elements and public PSTN numbers are not revealed Licensing and Capacity Use the display system-parameters customer-options command to verify that the Maximum Administered SIP Trunks value on Page 2 is sufficient to support the desired number of simultaneous SIP calls across all SIP trunks at the enterprise including any trunks to the service provider. The example shows that 800 SIP trunks are available and 208 are in use. The license file installed on the system controls the maximum values for these attributes. If a required feature is not or there is insufficient capacity, contact an authorized Avaya sales representative to add additional capacity. display system-parameters customer-options Page 2 of 10 OPTIONAL FEATURES IP PORT CAPACITIES USED Maximum Administered H.323 Trunks: Maximum Concurrently Registered IP Stations: Maximum Administered Remote Office Trunks: 0 0 Maximum Concurrently Registered Remote Office Stations: 0 0 Maximum Concurrently Registered IP econs: 0 0 Max Concur Registered Unauthenticated H.323 Stations: 0 0 Maximum Video Capable H.323 Stations: 0 0 Maximum Video Capable IP Softphones: 0 0 Maximum Administered SIP Trunks: Maximum Administered Ad-hoc Video Conferencing Ports: 0 0 Maximum Number of DS1 Boards with Echo Cancellation: 0 0 Maximum TN2501 VAL Boards: 10 1 Maximum Media Gateway VAL Sources: 0 0 Maximum TN2602 Boards with 80 VoIP Channels: Maximum TN2602 Boards with 320 VoIP Channels: Maximum Number of Expanded Meet-me Conference Ports: of 60

9 5.2. Special Application SA8965 TELUS requires that all INVITE messages contain SDP information, including re-invites. In general, when Communication Manager sends a re-invite to perform a media shuffling operation (redirect media directly between two endpoints) the re-invite will not include SDP information. In order to change this behavior, special application SA8965 must be. This is done via the change system-parameters special-applications command. Navigate to Page 7 and enter a y next to the special application titled SA SIP Shuffling with SDP in the list below. By enabling this feature, a new protocol variation parameter will appear on Page 3 of the trunk form (See Section 5.8). change system-parameters special-applications Page 7 of 9 SPECIAL APPLICATIONS (SA8888) - Per Station Music On Hold? n (SA8889) - Verizon VoiceGenie SIP MIME Message Bodies? n (SA8891) - Verizon VoiceGenie SIP Headers? n (SA8893) - Blast Conference? n (SA8896) - IP Softphone Lamp Control? n (SA8900) - Support for NTT Call Screening? n (SA8904) - Location Based Call Type Analysis? n (SA8911) - Expanded Public Unknown Table? n (SA8917) - LSP Redirect using special coverage point? n (SA8927) - Increase Paging Groups? n (SA8928) - Display Names on Bridged Appearance Labels? n (SA8931) - Send IE with EC500 Extension Number? n (SA8942) - Multiple Unicode Message File Support? n (SA8944) - Multiple Logins for Single IP Address? n (SA8946) - Site Data Expansion? n (SA8958) - Increase BSR Polling/Interflow Pairs to 40000? n (SA8965) - SIP Shuffling with SDP? y (SA8967) - Mask CLI and Station Name for QSIG/ISDN Calls? y 5.3. System Features Use the change system-parameters features command to set the Trunk-to-Trunk Transfer field to all to allow incoming calls from the PSTN to be transferred to another PSTN endpoint. If for security reasons, incoming calls should not be allowed to transfer back to the PSTN then leave the field set to none. change system-parameters features Page 1 of 19 FEATURE-RELATED SYSTEM PARAMETERS Self Station Display Enabled? n Trunk-to-Trunk Transfer: all Automatic Callback with Called Party Queuing? n Automatic Callback - No Answer Timeout Interval (rings): 3 Call Park Timeout Interval (minutes): 10 Off-Premises Tone Detect Timeout Interval (seconds): 20 AAR/ARS Dial Tone Required? y 9 of 60

10 On Page 9, verify that a text string has been defined to replace the Calling Party Number (CPN) for restricted or unavailable calls. This text string is entered in the two fields highlighted below. The compliance test used the value of unknown for both. change system-parameters features Page 9 of 19 FEATURE-RELATED SYSTEM PARAMETERS CPN/ANI/ICLID PARAMETERS CPN/ANI/ICLID Replacement for Restricted Calls: unknown CPN/ANI/ICLID Replacement for Unavailable Calls: unknown DISPLAY TEXT Identity When Bridging: principal User Guidance Display? n Extension only label for Team button on 96xx H.323 terminals? n INTERNATIONAL CALL ROUTING PARAMETERS Local Country Code: International Access Code: ENBLOC DIALING PARAMETERS Enable Enbloc Dialing without ARS FAC? n CALLER ID ON CALL WAITING PARAMETERS Caller ID on Call Waiting Delay Timer (msec): of 60

11 5.4. IP Node Names Use the change node-s ip command to verify that node s have been previously defined for the IP addresses of the CLAN circuit pack (clan1) and for the 3800 Net-Net SBC (spacme1). These node s will be needed for defining the service provider signaling group in Section 5.7. change node-s ip Page 1 of 2 IP NODE NAMES Name IP Address bvsm clan default medpro procr... procr procr spacme Codecs Use the change ip-codec-set command to define a list of codecs to use for calls between the enterprise and the service provider. G.711MU is the preferred/default codec used by TELUS. G.729 is supported but is not available on all media gateways in the TELUS network. Thus, the list of offered codecs must include G.711Mu. For the compliance test, G.729A, G.711Mu and G.711A were defined in IP codec set 4. To use these codecs, enter G.729A, G.711MU and G.711A in the Audio Codec column of the table in the order of preference. Default values can be used for all other fields. change ip-codec-set 4 Page 1 of 2 Codec Set: 4 IP Codec Set Audio Silence Frames Packet Codec Suppression Per Pkt Size(ms) 1: G.729A n : G.711MU n : G.711A n of 60

12 On Page 2, to enable T.38 fax, set the Fax Mode to t.38-standard. Otherwise, set the Fax Mode to off. change ip-codec-set 4 Page 2 of 2 IP Codec Set Allow Direct-IP Multimedia? n Mode Redundancy FAX t.38-standard 0 Modem off 0 TDD/TTY US 3 Clear-channel n 0 12 of 60

13 5.6. IP Network Region Create a separate IP network region for the service provider trunk. This allows for separate codec or quality of service settings to be used (if necessary) for calls between the enterprise and the service provider versus calls within the enterprise or elsewhere. For the compliance test, IP network region 4 was chosen for the service provider trunk. Use the change ip-network-region 4 command to configure region 4 with the following parameters: Set the Authoritative Domain field to match the SIP domain of the enterprise. In this configuration, the domain is avaya.com. This appears in the From header of SIP messages originating from this IP region. Enter a descriptive in the Name field. Enable IP-IP Direct Audio (shuffling) to allow audio traffic to be sent directly between IP endpoints without using media resources in the Avaya Media Gateway. Set both Intra-region and Inter-region IP-IP Direct Audio to yes. This is the default setting. Shuffling can be further restricted at the trunk level on the Signaling Group form. Set the Codec Set field to the IP codec set defined in Section 5.5. Default values can be used for all other fields. change ip-network-region 4 Page 1 of 19 IP NETWORK REGION Region: 4 Location: Authoritative Domain: avaya.com Name: SP Region MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes Codec Set: 4 Inter-region IP-IP Direct Audio: yes UDP Port Min: 2048 IP Audio Hairpinning? n UDP Port Max: 3329 DIFFSERV/TOS PARAMETERS RTCP Reporting Enabled? y Call Control PHB Value: 46 RTCP MONITOR SERVER PARAMETERS Audio PHB Value: 46 Use Default Server Parameters? y Video PHB Value: P/Q PARAMETERS Call Control 802.1p Priority: 6 Audio 802.1p Priority: 6 Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS H.323 IP ENDPOINTS RSVP Enabled? n H.323 Link Bounce Recovery? y Idle Traffic Interval (sec): 20 Keep-Alive Interval (sec): 5 Keep-Alive Count: 5 13 of 60

14 On Page 3, define the IP codec set to be used for traffic between region 4 and region 1. Enter the desired IP codec set in the codec set column of the row with destination region (dst rgn) 1. Default values may be used for all other fields. The example below shows the settings used for the compliance test. It indicates that codec set 4 will be used for calls between region 4 (the service provider region) and region 1 (the rest of the enterprise). Creating this table entry for IP network region 4 will automatically create a complementary table entry on the IP network region 1 form for destination region 4. This complementary table entry can be viewed using the display ip-network-region 1 command and navigating to Page 4 (not shown). change ip-network-region 4 Page 3 of 19 Source Region: 4 Inter Network Region Connection Management I M G A t dst codec direct WAN-BW-limits Video Intervening Dyn A G c rgn set WAN Units Total Norm Prio Shr Regions CAC R L e 1 4 y NoLimit n t all 5.7. Signaling Group Use the add signaling-group command to create a signaling group between Communication Manager and the 3800 Net-Net SBC for use by the service provider trunk. This signaling group is used for inbound and outbound calls between the service provider and the enterprise. For the compliance test, signaling group 36 was used and was configured using the parameters highlighted below. Set the Group Type field to sip. Set the Transport Method to tcp. The transport method specified here is used between Communication Manager and 3800 Net-Net SBC. Set the IMS Enabled field to n. Set the Near-end Node Name to clan1. This node maps to the IP address of the CLAN circuit pack as defined in Section 5.4. Set the Far-end Node Name to spacme1. This node maps to the IP address of 3800 Net-Net SBC as defined in Section 5.4. Set the Near-end Listen Port and Far-end Listen Port to This is the standard port value for SIP over TCP. Set the Far-end Network Region to the IP network region defined for the service provider in Section 5.6. Set the Far-end Domain to the IP address of the TELUS SIP proxy. Set Direct IP-IP Audio Connections to y. This field will enable media shuffling on the SIP trunk allowing Communication Manager to redirect media traffic directly between the SIP trunk and the enterprise endpoint. If this value is set to n, then the Avaya Media Gateway will remain in the media path of all calls between the SIP trunk and the endpoint. Depending on the number of media resources available in the Avaya Media 14 of 60

15 Gateway, these resources may be depleted during high call volume preventing additional calls from completing. Set the DTMF over IP field to rtp-payload. This value enables Communication Manager to send DTMF transmissions using RFC Set the Alternate Route Timer to 15. This defines the number of seconds that Communication Manager will wait for a response (other than 100 Trying) to an outbound INVITE before selecting another route. If an alternate route is not defined, then the call is cancelled after this interval. Default values may be used for all other fields. add signaling-group 36 Page 1 of 1 SIGNALING GROUP Group Number: 36 IMS Enabled? n Group Type: sip Transport Method: tcp Near-end Node Name: clan1 Far-end Node Name: spacme1 Near-end Listen Port: 5060 Far-end Listen Port: 5060 Far-end Network Region: 4 Far-end Domain: Bypass If IP Threshold Exceeded? n Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y Session Establishment Timer(min): 3 IP Audio Hairpinning? n Enable Layer 3 Test? n Direct IP-IP Early Media? n H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 15 In order to allow inbound calls with CPN block (see Section 2.2 for details), a second signaling group (signaling group 37) was created with the Far-end Domain set to anonymous.invalid. All other settings should be the same as signaling group 36 above. add signaling-group 37 Page 1 of 1 SIGNALING GROUP Group Number: 37 IMS Enabled? n Group Type: sip Transport Method: tcp Near-end Node Name: clan1 Far-end Node Name: spacme1 Near-end Listen Port: 5060 Far-end Listen Port: 5060 Far-end Network Region: 4 Far-end Domain: anonymous.invalid Bypass If IP Threshold Exceeded? n Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y Session Establishment Timer(min): 3 IP Audio Hairpinning? n Enable Layer 3 Test? n Direct IP-IP Early Media? n H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): of 60

16 5.8. Trunk Group Use the add trunk-group command to create a trunk group for each of the signaling groups created in Section 5.7. For the compliance test, the first trunk group (trunk group 36) was configured using the parameters highlighted below. Set the Group Type field to sip. Enter a descriptive for the Group Name. Enter an available trunk access code (TAC) that is consistent with the existing dial plan in the TAC field. Set the Service Type field to public-ntwrk. Set the Signaling Group to the signaling group 36 shown in Section 5.7. Set the Number of Members field to the number of trunk members in the SIP trunk group. This value determines how many simultaneous SIP calls can be supported by this trunk. Default values were used for all other fields. add trunk-group 36 Page 1 of 21 TRUNK GROUP Group Number: 36 Group Type: sip CDR Reports: y Group Name: DirectToSPAcme1 COR: 1 TN: 1 TAC: 136 Direction: two-way Outgoing Display? n Dial Access? n Night Service: Queue Length: 0 Service Type: public-ntwrk Auth Code? n Signaling Group: 36 Number of Members: of 60

17 On Page 2, the Redirect On OPTIM Failure value is the amount of time (in milliseconds) that Communication Manager will wait for a response (other than 100 Trying) to a pending INVITE sent to an EC500 remote endpoint before selecting another route. If another route is not defined, then the call is cancelled after this interval. This time interval should be set to a value equal to the Alternate Route Timer on the signaling group form described in Section 5.7. Verify that the Preferred Minimum Session Refresh Interval is set to a value acceptable to the service provider. This value defines the interval that re-invites must be sent to keep the active session alive. For the compliance test, the value of 900 seconds was used. add trunk-group 36 Page 2 of 21 Group Type: sip TRUNK PARAMETERS Unicode Name: auto Redirect On OPTIM Failure: SCCAN? n Digital Loss Group: 18 Preferred Minimum Session Refresh Interval(sec): 900 On Page 3, set the Numbering Format field to public. This field specifies the format of the calling party number (CPN) sent to the far-end. Set the Replace Restricted Numbers and Replace Unavailable Numbers fields to y. This will allow the CPN displayed on local endpoints to be replaced with the value set in Section 5.3, if the inbound call CPN block. For outbound calls, these same settings request that CPN block be activated on the far-end destination if a local user requests CPN block on a particular call routed out this trunk. Default values were used for all other fields. add trunk-group 36 Page 3 of 21 TRUNK FEATURES ACA Assignment? n Measured: none Maintenance Tests? y Numbering Format: public UUI Treatment: service-provider Replace Restricted Numbers? y Replace Unavailable Numbers? y Show ANSWERED BY on Display? y 17 of 60

18 On Page 4, set the Network Call Redirection field to n. Set the Send Diversion Header field to y and the Support Request History field to n. The Send Diversion Header field provides additional information to the network if the call has been re-directed. These settings are needed by TELUS to support call forwarding of inbound calls back to the PSTN and some Extension to Cellular (EC500) call scenarios. Set the Telephone Event Payload Type to 101, the value preferred by TELUS. Set the Shuffling with SDP field to y. This will instruct Communication Manager to send SDP information in shuffling re-invites on calls that use this trunk. This parameter only appears if special application SA8965 is. See Section 5.2 for full details. add trunk-group 36 Page 4 of 21 PROTOCOL VARIATIONS Mark Users as Phone? n Prepend '+' to Calling Number? n Send Transferring Party Information? n Network Call Redirection? n Send Diversion Header? y Support Request History? n Telephone Event Payload Type: 101 Shuffling with SDP? y In order to allow inbound calls with CPN block (see Section 2.2 for details), a second trunk group (trunk group 37) was created for the signaling group 37 created in Section 5.7. Use the instructions for creating trunk group 36 to create trunk group 37. Use a unique value for the Group Name and TAC. Set the Direction to incoming so only inbound calls will be allowed on this trunk. Set the Signaling Group to 37. Page 1 of trunk group 37 is shown below. All other settings on all other pages are the same as trunk 36. add trunk-group 37 Page 1 of 21 TRUNK GROUP Group Number: 37 Group Type: sip CDR Reports: y Group Name: Inbound from Telus COR: 1 TN: 1 TAC: 137 Direction: incoming Outgoing Display? n Dial Access? n Night Service: Queue Length: 0 Service Type: public-ntwrk Auth Code? n Signaling Group: 37 Number of Members: of 60

19 5.9. Calling Party Information The calling party number is sent in the SIP From, Contact and PAI headers. Since public numbering was selected to define the format of this number (Section 5.8), use the change public-unknown-numbering command to create an entry for each extension which has a DID assigned. The DID number will be assigned by the SIP service provider. It is used to authenticate the caller. In the sample configuration, multiple DID numbers were assigned for testing. These numbers were assigned to extensions 30023, and Thus, these same 10-digit numbers were used in the outbound calling party information on the service provider trunk when calls were originated from these extensions. change public-unknown-numbering 0 Page 1 of 1 NUMBERING - PUBLIC/UNKNOWN FORMAT Total Ext Ext Trk CPN CPN Len Code Grp(s) Prefix Len Total Administered: Maximum Entries: In a real customer environment, normally the DID number is comprised of the local extension plus a prefix. If this is true, then a single public-unknown-numbering entry can be applied for all extensions. In the example below, all stations with a 5-digit extension beginning with 3 will send the calling party number as the CPN Prefix plus the extension number. change public-unknown-numbering 0 Page 1 of 1 NUMBERING - PUBLIC/UNKNOWN FORMAT Total Ext Ext Trk CPN CPN Len Code Grp(s) Prefix Len Total Administered: Maximum Entries: of 60

20 5.10. Outbound Routing In these Application Notes, the Automatic Route Selection (ARS) feature is used to route outbound calls via the SIP trunk to the service provider. In the sample configuration, the single digit 9 is used as the ARS access code. Enterprise callers will dial 9 to reach an outside line. This common configuration is illustrated below with little elaboration. Use the change dialplan analysis command to define a dialed string beginning with 9 of length 1 as a feature access code (fac). change dialplan analysis Page 1 of 12 DIAL PLAN ANALYSIS TABLE Location: all Percent Full: 2 Dialed Total Call Dialed Total Call Dialed Total Call String Length Type String Length Type String Length Type 1 3 dac 2 5 ext aar 3 5 ext ext 4 5 ext 5 5 ext 6 5 ext 7 7 ext 8 1 fac 9 1 fac * 3 fac # 3 fac Use the change feature-access-codes command to configure 9 as the Auto Route Selection (ARS) Access Code 1. change feature-access-codes Page 1 of 8 FEATURE ACCESS CODE (FAC) Abbreviated Dialing List1 Access Code: *01 Abbreviated Dialing List2 Access Code: *02 Abbreviated Dialing List3 Access Code: *03 Abbreviated Dial - Prgm Group List Access Code: *04 Announcement Access Code: *05 Answer Back Access Code: Attendant Access Code: Auto Alternate Routing (AAR) Access Code: 8 Auto Route Selection (ARS) - Access Code 1: 9 Access Code 2: Automatic Callback Activation: Deactivation: Call Forwarding Activation Busy/DA: *13 All: *11 Deactivation: *12 20 of 60

21 Use the change ars analysis command to configure the routing of dialed digits following the first digit 9. The example below shows a subset of the dialed strings tested as part of the compliance test. See Section 2.1 for the complete list of call types tested. All dialed strings are mapped to route pattern 36 which contains the SIP trunk to the service provider (as defined next). change ars analysis 0 Page 1 of 2 ARS DIGIT ANALYSIS TABLE Location: all Percent Full: 2 Dialed Total Route Call Node ANI String Min Max Pattern Type Num Reqd op n op n intl n fpna n fpna n fpna n fpna n fpna n 21 of 60

22 The route pattern defines which trunk group will be used for the call and performs any necessary digit manipulation. Use the change route-pattern command to configure the parameters for the service provider route pattern in the following manner. The example below shows the values used for route pattern 36 during the compliance test. Pattern Name: Enter a descriptive. Grp No: Enter the outbound trunk group for the SIP service provider. For the compliance test, trunk group 36 was used. FRL: Set the Facility Restriction Level (FRL) field to a level that allows access to this trunk for all users that require it. The value of 0 is the least restrictive level. Pfx Mrk: 1 The prefix mark (Pfx Mrk) of one will prefix any FNPA 10-digit number with a 1 and leave numbers of any other length unchanged. This will ensure digits are sent to the service provider for long distance North American Numbering Plan (NANP) numbers. LAR: next change route-pattern 36 Page 1 of 3 Pattern Number: 34 Pattern Name: AcmeSBC Route SCCAN? n Secure SIP? n Grp FRL NPA Pfx Hop Toll No. Inserted DCS/ IXC No Mrk Lmt List Del Digits QSIG Dgts Intw 1: n user 2: n user 3: n user 4: n user 5: n user 6: n user BCC VALUE TSC CA-TSC ITC BCIE Service/Feature PARM No. Numbering LAR M 4 W Request Dgts Format Subaddress 1: y y y y y n n rest next 2: y y y y y n n rest none 3: y y y y y n n rest none 4: y y y y y n n rest none 5: y y y y y n n rest none 6: y y y y y n n rest none 22 of 60

23 6. Configure Acme Packet 3800 Net-Net Session Border Controller The following sections describe the provisioning of the Acme Packet 3800 Net-Net SBC. Only the Acme Packet provisioning required for the reference configuration is described in these Application Notes. The resulting SBC configuration file is shown in Appendix A. The Acme Packet SBC was configured using the Acme Packet CLI via a serial console port connection. An IP remote connection to a management port is also supported. The following are the generic steps for configuring various elements. 1. Log in with the appropriate credentials. 2. Enable the Superuser mode by entering enable and the appropriate password (prompt will end with #). 3. In Superuser mode, type configure terminal and press <ENTER>. The prompt will change to (configure)#. 4. Type the of the element that will be configured (e.g., session-router). 5. Type the of the sub-element, if any (e.g., session-agent). 6. Type the of the parameter followed by its value (e.g., ip-address). 7. Type done. 8. Type exit to return to the previous menu. 9. Repeat steps 4-8 to configure all the elements. When finished, exit from the configuration mode by typing exit until returned to the Superuser prompt. 10. Type save-configuration to save the configuration. 11. Type activate-configuration to activate the configuration. Once the provisioning is complete, the configuration may be verified by entering the show running-config command Physical Interfaces This section defines the physical interfaces to the private enterprise and public networks Public Interface Create a phy-interface to the public side of the Acme. 1. Enter system phy-interface 2. Enter s0p0 3. Enter operation-type Media 4. Enter port 0 5. Enter slot 0 6. Enter duplex-mode FULL 7. Enter speed Enter done 9. Enter exit 23 of 60

24 Private Interface Create a phy-interface to the private enterprise side of the Acme. 1. Enter system phy-interface 2. Enter s1p0 3. Enter operation-type Media 4. Enter port 0 5. Enter slot 1 6. virtual-mac 00:08:25:a0:f4:8a Virtual MAC addresses are assigned based on the MAC address assigned to the Acme. This MAC address is found by entering the command show prom-info mainboard (e.g a0 fa 80). To define a virtual MAC address, replace the last digit with 8 thru f. 7. Enter duplex-mode FULL 8. Enter speed Enter done 10. Enter exit 6.2. Network Interfaces This section defines the network interfaces to the private enterprise and public IP networks Public Interface Create a network-interface to the public side of the Acme. The compliance test was performed with a direct Internet connection to the service using the settings below. 1. Enter system network-interface 2. Enter s0p0 3. Enter ip-address Enter netmask Enter gateway Enter dns-ip-primary Enter hip-ip-list Enter icmp-ip-list Enter done 10. Enter exit Private Interface Create a network-interface to the private enterprise side of the Acme. 1. Enter system network-interface 2. Enter s1p0 3. Enter ip-address Enter netmask Enter gateway Enter hip-ip-list of 60

25 7. Enter icmp-ip-list Enter done 9. Enter exit 6.3. Realms Realms are used as a basis for determining egress and ingress associations between physical and network interfaces as well as applying header manipulation such as NAT Outside Realm Create a realm for the external network. 1. Enter media-manager realm-config 2. Enter identifier EXTERNAL 3. Enter network-interfaces s0p0:0 4. Enter done 5. Enter exit Inside Realm Create a realm for the internal network. 1. Enter media-manager realm-config 2. Enter identifier INTERNAL2 3. Enter network-interfaces s1p0:0 4. Enter done 5. Enter exit 6.4. Steering-Pools Steering pools define sets of ports that are used for steering media flows thru the 3800 Net-Net SBC Outside Steering-Pool Create a steering-pool for the outside network. The start-port and end-port values should specify a range acceptable to the service provider. For the compliance test, no specific range was specified by the service provider, so the start and end ports shown below were chosen arbitrarily. 1. Enter media-manager steering-pool 2. Enter ip-address Enter start-port Enter end-port Enter realm-id EXTERNAL 6. Enter done 7. Enter exit 25 of 60

26 Inside Steering-Pool Create a steering-pool for the inside network. The start-port and end-port values should specify a range acceptable to the internal enterprise network and include the port range used by Communication Manager. For the compliance test, a wide range was selected that included the default port range that Communication Manager uses and shown on the ip-network-region form in Section Enter media-manager steering-pool 2. Enter ip-address Enter start-port Enter end-port Enter realm-id INTERNAL2 6. Enter done 7. Enter exit 6.5. Media-Manager Verify that the media-manager process is. 1. Enter media-manager media-manager 2. Enter select show Verify that the media-manager state is. If not, perform steps Enter state 4. Enter done 5. Enter exit 6.6. SIP Configuration This command sets the values for the 3800 Net-Net SBC SIP operating parameters. The homerealm is the internal default realm for the 3800 Net-Net SBC and the egress-realm is the realm that will be used to send a request if a realm is not specified elsewhere. If the egress-realm is blank, the home-realm is used instead. 1. Enter session-router sip-config 2. Enter state 3. Enter operation-mode dialog 4. Enter home-realm-id INTERNAL2 5. Enter egress-realm-id 6. Enter nat-mode Public 7. Enter done 8. Enter exit 6.7. SIP Interfaces The SIP interface defines the SIP signaling interface (IP address and port) on the 3800 Net-Net SBC. 26 of 60

27 Outside SIP Interface Create a sip-interface for the outside network. 1. Enter session-router sip-interface 2. Enter state 3. Enter realm-id EXTERNAL 4. Enter sip-port a. Enter address b. Enter port 5060 c. Enter transport-protocol UDP d. Enter allow-anonymous agents-only e. Enter done f. Enter exit 5. Enter stop-recurse 401, Enter done 7. Enter exit Inside SIP Interface Create a sip-interface for the inside network. 1. Enter session-router sip-interface 2. Enter state 3. Enter realm-id INTERNAL2 4. Enter sip-port a. Enter address b. Enter port 5060 c. Enter transport-protocol TCP d. Enter allow-anonymous all e. Enter done f. Enter exit 5. Enter stop-recurse 401, Enter done 7. Enter exit 6.8. Session-Agents A session-agent defines an internal next hop signaling entity for the SIP traffic. A realm is associated with a session-agent to identify sessions coming from or going to the session-agent. A session-agent is defined for the service provider (outside) and Communication Manager CLAN (inside). SIP header manipulations can be applied to the session-agent level Outside Session-Agent Create a session-agent for the outside network. 1. Enter session-router session-agent 2. Enter host of 60

28 3. Enter ip-address Enter port Enter state 6. Enter app-protocol SIP 7. Enter transport-method UDP 8. Enter realm-id EXTERNAL 9. Enter description TELUS 10. Enter ping-method OPTIONS;hops=0 11. Enter ping-interval Enter ping-send-mode keep-alive 13. Enter in-manipulationid 14. Enter out-manipulationid outmantosp 15. Enter done 16. Enter exit Inside Session-Agent Create a session-agent for the inside network. 1. Enter session-router session-agent 2. Enter host Enter ip-address Enter port Enter state 6. Enter app-protocol SIP 7. Enter transport-method StaticTCP 8. Enter realm-id INTERNAL2 9. Enter description TrentonClan1 10. Enter ping-method 11. Enter ping-interval Enter ping-send-mode keep-alive 13. Enter in-manipulationid inmanfromcm 14. Enter done 15. Enter exit 6.9. Local Policies Local policies allow SIP requests from the INTERNAL2 realm to be routed to the service provider session agent in the EXTERNAL realm (and vice-versa) INTERNAL2 to EXTERNAL Create a local-policy for the INSIDE realm. 1. Enter session-router local-policy 2. Enter from-address * 3. Enter to-address * 4. Enter source-realm INTERNAL2 28 of 60

29 5. Enter state 6. Enter policy-attributes a. Enter next-hop b. Enter realm EXTERNAL c. Enter terminate-recursion d. Enter app-protocol SIP e. Enter state f. Enter done g. Enter exit 7. Enter done 8. Enter exit EXTERNAL to INTERNAL2 Create a local-policy for the EXTERNAL realm. 1. Enter session-router local-policy 2. Enter from-address * 3. Enter to-address * 4. Enter source-realm EXTERNAL 5. Enter state 6. Enter policy-attributes a. Enter next-hop b. Enter realm INTERNAL2 c. Enter terminate-recursion d. Enter app-protocol SIP e. Enter state f. Enter done g. Enter exit 7. Enter done 8. Enter exit SIP Manipulations SIP manipulation specifies rules for manipulating the contents of specified SIP headers. Two separate sets of SIP manipulations were required for the compliance test listed below. inmanfromcm A set of SIP header manipulation rules (HMRs) on traffic from Communication Manager to the SBC. outmantosp - A set of SIP header manipulation rules on traffic from the SBC to service provider (TELUS) Communication Manager to SBC The following set of SIP HMRs is applied to traffic from Communication Manager to the SBC. In some call flows the user part of the SIP Contact header received from Communication Manager was not passed unaltered to the public side of the SBC. To correct this, the user part of the Contact header is stored when received from Communication Manager and used to create a 29 of 60

30 temporary header called X-Contact that will be deleted on the outbound (public) side of the SBC. The information contained in the X-Contact header will be used to recreate the proper Contact header on the public side of the SBC as shown in Sections and To create this set of SIP HMRs: 1. Enter session-router sip-manipulation 2. Enter inmanfromcm 3. Enter description Inbound SIP HMRs From CM 4. Proceed to the following sections. Once all sections are completed then proceed with Steps 5 and 6 below. 5. Enter done 6. Enter exit Store Contact This rule stores the user part of the incoming Contact header. 1. Enter header-rule 2. Enter strcon 3. Enter header- Contact 4. Enter manipulate 5. Enter comparison-type case-sensitive 6. Enter msg-type request 7. Enter methods INVITE,UPDATE 8. Enter element-rule a. Enter strval b. Enter type uri-user c. Enter store d. Enter match-val-type any e. Enter comparison-type case-sensitive f. Enter match-value (.*) g. Enter done h. Enter exit 9. Enter done 10. Enter exit Create X-Contact This rule creates a temporary header called X-Contact containing only the user part of the incoming Contact header as stored by the rule defined in the previous section. 1. Enter header-rule 2. Enter addxcontact 3. Enter header- X-Contact 4. Enter add 5. Enter comparison-type pattern-rule 30 of 60